AF’s Weblog

June 1, 2012

How to Get the Pumping Drums Effect with Sidechain Compression

To read the full article see:  Sidechain Compression

Sidechaining has been around for years; this is the process of using one signal to control another. A couple classic examples are using a kick drum to gate a bass part, or doing de-essing – isolating the high sibilant frequencies from a vocal, and using those to trigger compression so that the sibilants come down in volume.

But in the digital age, we can do a lot more with sidechaining. One of the most popular applications is with dance music, where sidechaining can create the “heavy pumping” electronica drum sound used by artists like Eric Prydz and others.

We’ll describe how to do this with Cakewalk Sonar, although the same principle applies to other programs that allow for sidechaining. Sonar allows sidechaining for several effects, including compression, so that one instrument can control the compression characteristics of another instrument. This offers a variety of effects, including a “pumping” drum sound for multitracked drum parts; we’ll do that by setting up the snare to control compression for all drum tracks.

Fig. 1: You’ll need a drum submix bus to create an overall drum sound.

The first step is to create a drum submix bus, and send the drum tracks to it (Fig. 1). We need this submix so the entire drum track can be processed by the sidechained compressor. To create the submix bus, right-click in an empty space in the bus pane and select “Insert Stereo Bus.” To create a send in track view, right-click in a blank space in the track title bar and select “Insert Send.” From the menu that appears, select the send destination. Make sure you feed the bus pre-fader, and turn the individual drum channel faders down so that only the bus contributes the drum sound to the master.

Fig. 2: Assign the Drum Submix out to your main stereo output.

Let’s take a closer look…

….

Create a second pre-fader send in the snare track, and assign its out to the bus feeding the sidechain input.

Fig. 7: We’re almost there – it’s time to adjust the compressor.

To adjust the compressor, start with the compression attack time set to 0 ms; the drum sound will essentially disappear when the snare hits because the gain is being reduced so much. Gradually increase the attack time to let through more of the initial snare hit, and add a fair amount of release (250-500 ms) to increase the apparent amount of pumping.

And there you have it – the pumping drum sound. May it go over well on the dance floor!

To read the full article see:  Sidechain Compression

October 4, 2011

IK Multimedia T-RackS Black 76 & T-RackS White 2A and Native Instruments VC 76 & VC 2A Review

The (almost) simultaneous launch of the 1176 and LA2A software versions by IK Multimedia and Native Instruments is a good opportunity to make a quick comparison. Let’s go!

Some hardware products —instruments, signal and effect processors— have a kind of Holy Grail status. Among studio processors —and regardless of their denomination: limiting amplifiers, leveling amplifiers or just compressors— the Teletronix LA-2A and Urei/Universal Audio 1176 LN, as well as the legendary Fairchild 660 & 670, which are extremely rare to find (we saw a unit sold for $42,000 on ebay…), are highly regarded pieces of gear you’ll still find in big studios either as original or reissue versions. Many recording studios also use more or less faithful replicas of the originals designed by manufacturers ranging from Studio Electronics to Purple Action that (mostly) have a great sound quality. It’s simple: these legendary tools can be heard on almost every album ever produced since they were first introduced.

As expected, the software world also has its own interpretation of these legends. Since the early attempts by Bomb Factory to the latest products by IK Multimedia and Native Instruments in collaboration with Softube, and including the existing Universal Audio, URS and Waves products, the market is packed with software simulations.

We do not intend to review all software versions (they are too many) nor to compare them with vintage or modern hardware products. We just want to compare two manufacturers and use the same plugins for Universal Audio’s platform UAD-1/2 as a reference, without neglecting the performance of the original processors and some hardware replicas. Note that Native Instruments’ Vintage Compressors bundle also includes the famous dbx 160 compressor, which we won’t take into account for this review.

Introducing the Plug-ins

Test system

MacPro Xeon 3.2 GHz

OS 10.6.7

Logic 9.1.4

IK Multimedia T-RackS3 Black 76 and White 2A v.3.5.1

Native Instrument Vintage Compressors VC 76 and VC 2A

UAD-2 UAD 1176 LN and UAD LA-2A v.5.9.1

Native Instruments decided to collaborate with Softube, a manufacturer that has launched quite exceptional plugins — I can’t seem to get enough of — like the Acoustic Feedback and the Tubetech CL 1B. I haven’t had the opportunity to try out their reverbs and amp simulations yet, but other user’s feedback is very promising. So let’s start with the Urei 1176 LN and Teletronix LA-2A emulations (VC 76 & VC 2A) conceived for Guitar Rig 4, like the other Studio Effects of the manufacturer. The good news is that the Guitar Rig 4 player is free. The bad news is that you can’t use the plugins unless you have the manufacturer’s guitar multi-effect. Available for Mac (Intel only) and PC in 32-bit and 64-bit versions, the bundle —including the host (GR4 or GR4 Player) plus the plugins— supports AU, VST and RTAS formats and includes a standalone version. As always, activation is done via the Service Center.

As their name already implies, IK Multimedia’s T-RackS Black 76 and T-RackS White 2A were conceived to work within T-Racks 3, but also as individual 32-bit or 64-bit plugins to be used directly in any DAW that supports AU, VST or RTAS. As usual, to activate them you’ll have to use the Authorization Manager.

As for Universal Audio’s UAD 1176LN and UAD LA2A plugins, they are only available for the DSP cards developed by the manufacturer, from the UAD-1 to the UAD-2 Quad. Our card uses OS 5.9.1. The plugins are Mac and PC compatible, they support 32-bit and 64-bit operation (only the card drivers actually work in 64 bits, the plugins still operate at 32 bits and require a bridge), and work with AU, VST and RTAS.

Did you ask for the 1176…?

All software manufacturers took their inspiration from the LN version of the FET 1176 compressor. LN (for Low Noise) means that the product includes a modification made by Mr Plunkett, engineer at Urei, who wanted to reduce the noise. You can recognize it by the famous black front instead of the traditional burst aluminum front with blue stripes around the VU-meter. These versions, referred to as C, D and E, are the most venerated. From a technical and audio standpoint, FET compressors contributed the principle of adjustable ratio, as well as much shorter attack and release times compared to competitors using Vari-Mu or optical designs.

IK Multimedia designed its plugin based on an E model. Neither Native Instruments nor Softube give any information on this matter. However, Softube’s experience developing the FET Compressor was certainly an advantage for Native Instruments. UA doesn’t give any information about the model either. Anyway, all three have a very similar sound character, just like signal processors that use only analog components (don’t forget that every unit sounds slightly different than the other, even if it’s the exact same version).

As for the features, except for the major innovation in the form of a stereo version (we’re talking about the 1176, not the Urei 1178 stereo version), UA stays faithful to the original. Thus, you get all the original features users like so much for their ease of use: a pair of big controls for input and output adjustment, two smaller Attack and Release knobs, four Ratio buttons (4:1, 8:1, 12:1, and 20:1) including an All-Buttons mode, and four VU-meter buttons (Off, +4, +8, and GR). However, you don’t have the possibility to set the Attack control to Off, which would switch the compressor to ratio 1:1, meaning you could process the signal without compression in order to get only the device’s sound character. All three manufacturers reproduced the (confusing, at least in the beginning) operation of the controls: the fastest attack time is not hard left (1) but hard right (7). The same applies to the release. Attack times range from 20 to 800 microseconds (yes, micro!), while release times range from 50 to 1,100 milliseconds.

Native Instruments and IK Multimedia added some modifications. IK Multimedia added four snapshots, taken from T-RackS’s architecture, plus L/R and M/S buttons allowing the user to choose between one of the two operating modes (well done!). Three additional buttons (L, R and =, where L and R become M and S in M/S mode) allow the user to process the two channels of a signal separately or together. IK Multimedia’s version also displays the setting values, but they don’t quite match reality, at least for Attack and Release, which is a pity. What’s the use of adding values if they have no meaning… The All-Buttons mode is accessible via a dedicated knob. Ratio 1:1 is available clicking the Off button under the Attack control. You also have Bypass and Reset buttons.

In Native Instruments’ version, a Ratio slider replaces all four original buttons but offers all usual values (4:1 to 20:1 plus All-Buttons mode). The 1:1 ratio replaces the bypass button under the Attack control. The VU-meter management also changed: you can display the input level, output level and gain reduction. The plugin comes with a preset menu accessible via the advanced settings. The side-chain input (great for techno/electro fans) comes with a slider that allows the user to adjust the amount of direct signal so that parallel compression is possible by adjusting the output level (which has no effect on compression itself). Unlike Softube’s FET Compressor, note that you get no continuous Ratio setting.

Now let’s take a closer look…

Conclusion

So, what should you choose, Native Instruments or IK Multimedia (UA doesn’t count because it was only used as a reference)? It’s a difficult question because both options provide advantages and good sound results. Will these plugins replace real 1176LN and LA-2A hardware processors? No. Does their performance match the original hardware versions? Yes, except for some applications (All-Button mode, transients management in given situations, settings and/or VU-meter calibration). Is it possible to do a good job with them? Yes. Software manufacturers benefit from the ever-increasing computer performance and offer more authentic emulations every time. Take for example the 24/7, which was considered a really good plugin when it was launched…

And these tools are affordable, which is not the case of the hardware gear they are based on. Each IK Multimedia module is sold for €89.99 and can be used in T-Racks. The Native Instruments bundle (including the three compressors) is sold for €199, while single plugins go for €99; and Guitar Rig Player 4, which is required for their use, is free. Moreover, both manufacturers offer fully-usable demo versions, which is an excellent way for you to expand on this comparison.

To read the full detailed article see: Vintage Compressors Review

July 28, 2011

Mysteries of Dynamics Processing Revealed

Filed under: Compressors, Processors, Recording reviews — Tags: , , , , , — audiofanzine @ 7:34 pm

A dynamic processor is something that outputs a signal, where the level of the outgoing signal is based on the level of the incoming signal. In other words, a loud signal coming in will come out differently than a quiet signal coming in.

Basic types of Dynamic Processors

Compressors: The most common – the louder the signal is coming in, the less level it provides going out. In a compressor, a target level is set – called the “threshold” – and any signal coming in that exceeds that level will be reduced. The higher the level is above that threshold, the more reduction will occur. More on this later.

 

Limiters: Limiters are like super compressors. The idea is to ensure that the level does not exceed the threshold. Because this amount of compression is extreme, a limiter relies on certain functions and design that regular compressors do not have.

Expanders: The quieter the signal is coming in, the less level it provides going out. In other words – it makes quiet signals even quieter. Much like a compressor, the threshold is set at a certain level. Any signal that does NOT exceed that threshold is reduced, and the quieter the signal, the more reduction is done.

 

Gates: Gates are like super expanders. Anything that does not exceed the threshold is reduced to inaudible. Again, because gates are extreme, they often require a slightly different design than a regular expander.

 

Now – I’ll focus primarily on Compression, because that’s going to be the most commonly used dynamic processor.

Compression

Every signal you hear is compressed??? Yes, every signal you hear is compressed.

Bare with me. Imagine you have a rapper in front of a microphone. The rapper raps, you record. You play it back. You haven’t used any processing – you’re just playing back the raw vocals.  You are listening to a signal that has gone through at bare minimum 3 stages of compression – and more likely than not – closer to 6.

  • The microphone capsule gains tension as the rappers voice pushes it – in other words – it pushes back. The more the rapper’s voice pushes in – the harder the capsule diaphragm pushes back. In other words, the louder the signal is hitting the capsule, the more reduction the capsule does to the signal. That’s compression! (It’s mild compression, but it’s still compression).
  • Along the way through the microphone, you may hit a tube. Tubes have a non-linear response to voltage – the response is quite curved, and also changes the frequency balance of the signal. This is called saturation – which will tend to “round out” a signal, by reducing the loudest peaks. Compression! And before leaving the microphone, the signal may hit a transformer as well, which will saturate in a similar way… more compression.
  • The preamp is going to have multiple stages of saturation – and often times, the more gain you give something – the deeper that saturation curve goes. In other words, the more you drive the signal at the preamp, the more compression the signal experiences.
  • Then the sound has to actually come out of the speaker cones. Well, those speaker cones are going to build up tension when pushed further. See where this is going? This is called “cone compression”.

Ok – so this is a bit of a simplification – but there’s a point here. The point is that “compression” is always part of the signal. Some mics have less of it, some have more – same with speakers, tubes, transformers, etc. And they all do it in different ways. With tubes, people will talk about their saturation curves and %THD (total harmonic distortion – or frequency alterations). With mics, people will refer to how it “grabs” a sound – or more specifically – the sound’s shape.

Now let’s take a closer look…

Maximum Punch

There is a thin line between a transient sound, and a sustained sound. A sound that holds for any noticeable amount of time is sustaining. A sound that moves by too quickly to register as it’s own moment is transient. But transients can vary in length. A transient can be half a millisecond or it could also be ten milliseconds; they won’t sound the same. A big factor in punch is how long that transient exists. A quick transient sounds “spikey” – but a long transient sounds “punchy.” You want to find the point that makes the transient exist as long as possible before “flattening out” or becoming a sustained sound. Only your ear can tell you where that point is.

 

Good samples are already shaped to have that kind of impact – and any additional compression may actually soften that. Of course, punch has a lot to do with frequency as well – but that’s for another article.

 

Now what about the release? The release is super elusive. It determines how long it takes for the compressor to let go. If the release is too short for the signal you are going to get a disjointed sounding shape which usually results in distortion. If it’s too long, your signal never really returns to its natural shape, and you generally lose tone (or you just get permanent drive on the compressor’s output, giving the whole signal a new bit of tone). So the idea is to find a point that emphasizes the sustain (which is where most of the signals tone lives) properly.

 

Lastly, when the attack and release are set in a way that seem to argue – the compression can become very audible. You either hear the decent or the ascent of the signal level. This is called pumping. It’s generally annoying, but can sometimes be used an effect. If audibly desired, consider the rhythm of the release time, and ask yourself if it’s groove is complimenting the song.

———————————-

So, rather than thinking of a compressor as something that effects the “level” of a signal. Think of a compressor as something that effects shape. Why? Because level can be controlled with the volume fader more accurately and transparently. A fader doesn’t really control shape, unless you are being extremely meticulous. Conversely, compression will always effect the shape of the sound it is working on.

Once you start hearing shape, you will understand compression.

To read the full detailed article see:  Mysteries of Dynamics Processing Revealed

July 15, 2011

Vocals Processing Tips: Part 2

Hard disk recording techniques have affected every aspect of recording, including vocals. Although overdubbing vocals has been a common technique for years, today’s programs let you do multiple tracks of vocals, and make a “composite” with all the best bits. We’ll cover how to do that, then talk a bit about compression and reverb.

Composite Vocal Tracks

Cutting and pasting has benefited vocals, as you can do multiple takes, and splice the best parts together to make the perfect “composite” vocal. Some producers feel that stitching together vocals doesn’t produce as natural a “feel” as a take that goes all the way through from beginning to end, while others believe that being able to choose from multiple takes allows creating a vocal with more range than might occur with a single take. If you want to try composite vocals, here are the basic steps.

Record the Takes

Record enough takes so there’s plenty of material to piece together a good performance (loop recording is particularly handy for doing vocals). While you’re in a recording mood, record a little bit of the track without any input signal. This can be handy to have around, for reasons described later.

Audition the Takes

Audition each take, and isolate the good parts (by cutting out unwanted sections). I recommend setting loop points around very short phrases.

Solo each take, one after the other. If you’re not going to use a take, cut the phrase. If a take is a candidate for the final mix, keep it.

Pick the top 3 or 4 candidates, and remove the equivalent sections from the rest of the tracks. Now repeat this procedure, phrase by phrase, until you’ve gone over the entire performance and found the best bits

Ligne de chant compilée

In Sonar, several takes of vocals have been recorded. A mute tool has muted portions of each track (the waveforms are shown as shaded), with the remaining parts making up the final vocal.

Next, listen to combinations of the various different phrases. Balance technical and artistic considerations; choose parts that flow well together as well as sound technically correct. Sometimes you might deliberately choose a less expressive rendition of a line if it comes just before an emotional high point, thus heightening the contrast.

Once you have the segments needed for a cohesive performance, erase the unused parts. If you want to archive everything “just in case,” go for it. But if after putting the part together you think it could be better, you might be better off re-cutting it than putting more hours into editing.

Ligne de chant compiléeSeveral takes of vocals were recorded into Cubase SX, and edited to create one final vocal. The program shows the elements that make up the final vocal by highlighting them in green.

Bounce the Takes

This isn’t absolutely necessary, but converting all the bits into a single track simplifies subsequent editing and processing.

Before bouncing, play the tune through from start to finish and match the segment levels as closely as possible. Also check the meters for any send bus or master bus the tracks are feeding, and adjust levels (if needed) so there’s no distortion. Generally, the bounced track will be derived from a bus or master; if there’s distortion, the bounced track will have distortion too.

This is also where the recorded noise might come in handy. Sometimes I’ve had to do a quick fade on the end of one segment, and a fade in on the beginning of another, leaving a dead silent gap between phrases. Layering in a bit of the noise signal gives better continuity, and keeps the part from sounding too “assembled.”

After everything’s set, implement the program’s bounce or mix to hard disk function. You can typically bounce to an empty track, or “render” the audio to disk and bring it back into the project.

Edit the Composite Track

At this point, I bring the composite track into a digital audio editor for clean-up. Here are some typical processes:

  • Phrase-by-phrase gain adjustments. If a phrase has mismatched levels, use the program’s level change DSP or mix automation to fix the problem.
  • Fix breath noises and inhales. There might be “flammed” inhales from combining two different takes, so cut one. However, don’t eliminate all inhales and breath noises — they keep things “human.”
  • Add overall dynamics control, reverb, EQ, echo, etc. if needed. Do not add these while cutting individual takes; it will be much harder to match the effect, and in the case of reverb, tails might get cut off. Adding processing after optimizing the entire track will give the best results.

Tidy Up Your Hard Disk


After the vocals are done, check how your program deals with deleting unused segments, as this can reclaim significant space from your hard drive.

Now let’s take a look at compression…

Reverb Tips for Vocals

Nothing “gift wraps” a vocal better than some tasty reverb. My favorite reverb for voice is a natural acoustic space, but as reverb rooms are an endangered species, you’ll likely use a digital reverb. Reverb settings are a matter of taste, but two parameters are particularly important.

Waves RVerb (Renaissance Reverb)

A reverb’s Predelay and Diffusion parameters are crucial to getting good vocal sounds. This reverb, the RVerb plug-in from WAVES, offers an exceptional amount of control.

Diffusion: With vocals, I prefer low diffusion, where each reflection is more “separated.” Low diffusion settings often sound terrible with percussion, as the individual echoes can have an effect like marbles bouncing on a steel plate. But with vocals, the sparser amount of reflections prevent the voice from being overwhelmed by too “lush” a reverb sound.

Predelay: This works well in the 50-100 ms range. The delay allows the first part of the vocal to punch through without reverb, while the more sustained parts get the full benefit of the reverberated sound.

To read the full article see: Vocals Processing Tips Part 2

July 8, 2011

Vocals Processing Tips: Part 1

It was late at night, at a live-in-concert recording session in Germany. As several thousand fans waited anxiously, the vocalist walked onstage, and picked up a set of headphones. I saw him plug them into the mixer, and figured he was going to make one final check of his vocal sound before the band kicked in. He then turned the preamp gain control up full…not too unusual, as mics don’t have a lot of gain. But then he held the headphones up to his mouth and — started singing. He had plugged the headphones into the mic in, not the headphone out…and he had done it on purpose. Is this what recording vocals in the 21st century is about?

Well, the answer is yes…and no. No, in the sense that a well-recorded vocal through a high-end mic feeding a state-of-the-art preamp remains a supremely important part of the recording art. Yes, in the sense that it underscores a fundamental truth about recording today: anything goes.

The tools of the vocal trade have undergone as dramatic a transformation as the recording process itself. Microphones are better and cheaper; today’s “budget” mics sometimes outperform the champions of yesteryear. Preamps, whether tube or solid state, have noise levels that are measurable only with the most sensitive test equipment. Processing gear ranges from “vocal strips” dedicated solely to vocal, to technologies such as Antares’Auto-Tune (which can correct out-of-tune-vocals) and mic modeling, which mimics the characteristics of particular “signature” mics. Compressors, reverbs, even vocal booths have all enjoyed the results of technological progress.

So what’s the best way to record vocals these days? The answer, of course, is that anything goes. Following are some of the possibilities.

Recording Vocals

Few topics inspire more debate than the optimum vocal mic and preamp. But note that a mic and preamp combination that sounds great with one vocalist might not work with another. Case in point: once while recording, my voice was recorded with a sub-$100 dynamic mic and a $995 condenser mic. The unanimous agreement was that the dynamic sounded better.

Was it because the mic was better? No. From any objective standpoint, it was inferior. But it had some response anomalies that flattered my voice. The condenser mic was accurate, but my voice didn’t need accuracy: It needed a high-frequency lift, and warmth from the proximity effect (i.e., the tendency of a dynamic to produce more bass as you sing closer to it).

I sometimes wish that all mics looked the same, and had no labels on them. That would force engineers to take a fresh approach with every session. It’s very easy to rely on using old favorites — the assumption is that the mic that worked great on the last session will be equally good on the current session, but that isn’t always true. Furthermore, there’s a matching issue between mics and preamps, so mic X might sound great with preamp A and not so great with preamp B.

Bottom line: Try every mic with a vocalist, record the results, then choose which one sounds most appropriate. I suggest comparing two mics at a time to prevent “option overload.” Choose the best of each pair, then have a runoff among the winners.

Let’s take a look at some other tips…

Synthesizing Vocal Harmonies

Normally, I sing my own harmonies. But sometimes, pitch shifters — because they’re not perfect — add timbral and timing imperfections that actually sound better for some applications.

 

Here’s an example of creating harmonies using Sonar’s real-time pitch shifting plug-in (the principles are the same for other programs). Note that Sonar Producer Edition also includes a high-quality, but non-real-time, pitch stretch processor. I usually use the real-time plug-in to get the harmonies right, then go back and process the files destructively using the higher-quality, non-real-time algorithm.

Harmonisation d'une partie de chantThis shows harmonies being generated within Sonar using real-time plug-ins. Higher-quality, offline plug-ins can be used for the final processing.

Note that there are four tracks of vocals: The teal one at the top is the original vocal. The violet one below that is a “cloned” version, which has been processed with the doubling technique mentioned previously.

The next track (blue) is also a cloned track, but it’s being processed through the pitch shifter set to a major 3rd. However, note that some elements have been cut from this track and moved to the next track down, which is processed through the pitch shifter set to a minor 3rd. As Sonar doesn’t know which notes should receive minor 3rd or major 3rd harmonization, you have to cut up the track appropriately, and move the right phrases or notes to the right tracks. This may require zooming way in on the cloned track, so you can make cuts in the space between phrases.

The standard pitch shifting caution applies — the further you stretch pitch, the less realistic the sound. Sonar’s real-time pitch shifter does not preserve formants during shifts; however, when pitching up a major third the formant change adds a bit of voice-on-helium effect, which when mixed behind the main vocal, can actually sound pretty cool.

Starting with Sonar 5, the Producer Edition includes Roland’s VariPhrase technology in their V-Vocal plug-in. With this plug-in, you can “draw in” harmonics and constrain a melody to particular notes. This makes the process of harmonization much easier, as does a similar feature in Samplitude and Digital Performer. There are also programs like Antares Harmony Engine, and zplane’s Vielklang (among others) that are designed to generate harmonies.

To read the full detailed article see: Vocal Processing Tips Part 1

March 21, 2011

TC Helicon VoiceLive Touch Review

Vocal effects are present more than ever in modern music, but if you want special effects, you don’t give the control to the sound engineer. Loopers are also hip, especially for artists making solo performances on stage. So, when such a serious manufacturer as TC-Helicon offers a voice processor, harmonizer and looper in one single box (everything you need for a single voice, voice + guitar or voice + keyboards performance) it is worth taking a closer look at it. Let’s go!

Who is TC?

TC-Helicon is a Canadian company belonging to the TC-Group, a holding that controls several prestigious pro audio manufacturers like the very famous TC-Electronic, but also Lab.gruppen amps or Tannoy speakers… TC Group merged with Gibson in 2008. TC-Helicon specializes in voice gear, from processors to mics. Their products have a good overall quality and a rugged construction plus they sound pretty good.

The VoiceLive Touch is a variation of the VoiceLive 2 footboard, which offers more possibilities (to a certain extent) and is also 60% more expensive. The street price of the Voice Live 2 is around $800 while you can get the Voice Live Touch for only $500. And unlike the VoiceLive 2, the VoiceLive Touch includes a looper. And not a toy one.

The VoiceLive is not just a light version of its predecessor. In fact, its designers had the idea of developing a concept with a radically different user interface based on a “touch” interface. Definitely trendy, but is it a good thing? We’ll have to discuss the matter further. Let’s start discovering the unit!

Heavy Duty!

TC Helicon VoiceLive Touch

At first hand, the VoiceLive Touch gives the impression of being quite sturdy. Its heavy weight, in spite of the compact size, is responsible for that, but also the materials. The housing, except for the front panel with the touch interface, is made out of some sort of rugged rubber plastic. Why this material? Because the VoiceLive Touch’s special shape allows you to place it on a table or on a microphone stand, which is a very convenient solution to always have it at hand. The fixation system is well thought out as you can see from the pictures. However, the rear handle used to fix the unit to a microphone stand makes connections a bit harder, even if it hides the connectors from the audience. You can’t have everything, right? Long jacks can be problematic.

Although the overall design of the VoiceLive Touch is very nice and original, the touch interface looks a bit awkward, as if it had been designed in the 80′s. In some (rare) circumstances, you’ll be dazzled by the reflections. The external PSU is the same as for many consumer products, which is hard to understand for a stage device, especially at this price. An adapter plugged into a multi-outlet power strip or an extension power cord is not exactly what you want to have or see in the middle of the stage. Luckily, the length of the cable should be enough to hide it behind a monitor speaker. I know, it’s an insignificant detail, but once you discover its features you’ll agree with me that such a product deserves better.

An All-Rounder?

The features of the VoiceLive Touch are very comprehensive. Just take a look:

  • Voice effects processor
  • Automatic tuning correction (so you can sing in tune)
  • Harmonizer (to add a choir to your voice)
  • Basic guitar effect
  • Looper
  • Phrase sampler

Phew!

TC Helicon VoiceLive Touch

Let’s just mention the tuning corrector briefly: it offers a single “strength” control (in %). If you sing like an angel you’ll set it to 0 if, but if you are a shower singer you’ll probably have to set it to 100. As always with such tools, the performance looses its natural expression with higher settings. In extreme cases you’ll even experience bizarre results (like the famous “Cher” effect). But it’s a very handy tool: it is not easy to sing in tune with choirs and effects added to your own voice. The tuning corrector makes up for this. Since it is global, we prefer to use the one included with some effects for presets that require it. Do note that the samples were recorded without tuning correction (I guess you’ll hear it anyway!).

The processor includes six effect categories: harmonizer, modulation (chorus, flanger, …), delay, reverb, “double,” and “FX,” which includes different effects (only one can be used at a time). The effect chain has independent sections you can switch on/off individually (like separate stompboxes). Each section offers several (quite) basic parameters. Let’s take the delay as an example: you can choose from 18 different delay types, set the effect amount added to the mix, the stereo width, and the tempo. And that’s it. No direct feedback nor damping nor feedback delay time: just select the delay type number to change to a new sound. The 18 delays cover a wide range of effects but don’t allow the precise processing you get with standard parameters. Below you will see that that wasn’t the goal of the VoiceLive Touch.

TC Helicon VoiceLive Touch

You also get a “lead level” setting for each effects section. It allows you to attenuate your voice when only that effect is active. It is similar to a standard dry/wet setting but the fact that it activates only when no other effect is on allows you to create (very) interesting special effects.

Some sections offer more settings, but some are very basic. Besides the “lead level,” the reverb provides you only with a send level (routing to the mix) and a selector to choose among 30 reverb types, which is quite a lot considering that they all have different colors and duration. This approach is good because of its simplicity (some people don’t know what all parameters of a delay or a reverb are for), but it can be frustrating for people who are used to tweaking effects. Especially given that the effects are identified only by a number rather than a name, which makes it difficult to find them and often requires you to try all of them.

Thus, fine tuning your presets with the VoiceLive Touch demands quite some time before going on stage. Even though most factory presets sound very good, they require you to at least adjust levels.

Now let’s take a close look…

Conclusion

While I was very enthusiastic about the concept in the beginning, I ended up with mixed feelings because the VoiceLive Touch has some excellent features as well as some irritating ones. The touch interface didn’t quite convince me. In fact, I was surprised by some design faults, as well as by some very nice ideas and some complexities. But the VoiceLive Touch has many advantages too: besides its perfect sound quality, some very intelligent features and its versatility (I couldn’t mention many of its applications in this review), it has a very powerful harmonizer and an excellent looper. Both are crucial in the decision to buy the unit. When I was a solo performer (voice+guitar), I would have been delighted by such a product.

Advantages:

  • Original concept
  • Clever features
  • Irreproachable manufacturing quality
  • Very professional sound
  • Excellent harmonizer and effects
  • Awesome looper
  • Affordable optional footboard

Drawbacks:

  • Questionable “full touch” interface
  • No possibility to tweak effects live on stage
  • Improvable ergonomics
  • Lousy display
  • Settings sometimes too complex

To read the full multimedia article please see:  TC Helicon VoiceLive Touch Review

March 4, 2011

Noise Gates Don’t Have to be Boring

Noise gates aren’t as relevant as they were back in the analog days, when hiss was an uninvited intruder on anything you recorded. But noise gates can do some really cool special effects that have nothing to do with reducing hiss. This article shows how to make them a lot more interesting, and throws in a bunch of fun audio examples, too. But first, let’s do some noise gate basics for the uninitiated.

Noise Gates Basics

A noise gate mutes its output with low-level input signals, but higher-level signals can pass through. Following are the typical adjustable parameters found in a noise gate, whether analog, digital, or plug-in.

  • Threshold: If the input level to the gate passes below the threshold, the gate “closes” and mutes the output. Once the signal exceeds the threshold, the gate opens again.
  • Attack: This determines how long it takes for the gate to go from full off to full on once the input exceeds the threshold.
  • Decay: This sets the time required for the gate to go from full on to full off once the signal falls below the threshold. Since decaying signals often criss-cross the threshold as they decay, increasing the decay time prevents “chattering.”
  • Key input: Normally, the gate opens and closes based on the input signal’s amplitude. The key input allows patching in a different control signal for the gating action (for example, using a kick drum as the key signal to turn a sound on and off in time with the kick’s rhythm). Note that in most cases, you won’t find this in plug-ins, only in hardware units.

All right, let’s get into applications.

Selective Reverb

I was using a premixed drum loop from the Discrete Drums Series 2 library, but in one particular part of the song, I wished that the snare—and only the snare—had some reverb. Although Series 2 is a multitrack library, I didn’t want to go back and build up the drum loop from scratch. So why not just extract the snare drum sound, put some reverb on that, and mix it in with the drums?

Referring to Fig. 1, I copied the drum loop in Track 1 to a second track in my sequencer (if you were doing this in hardware, you’d split it into two mixer channel inputs). In the second track, there’s EQ inserted to roll off all the low end, which took most of the kick out of the signal, as well as the high end, to reduce the level of cymbal crashes.

Reverb sélective

Fig.1

The next step was to insert a noise gate in Track 2, and raise the gate threshold so that only the snare peaks made it through (the screen shot shows a Compressor/Gate plug-in, but the compressor was disabled by setting the ratio to 1:1). These peaks fed the reverb, which dumped into the master bus along with the original drums. The end result: Reverb on the snare only, added in with the rest of the drums.

Now let’s take a closer look and listen to some samples…

Kick Drum “Hum Drum”

Here’s a trick for hardware noise gates. Suppose you want to augment an existing kick drum sound with a monster rap kick, like that famous TR-808 rap sound. Here’s a sneaky way to do it:

  1. Set a sine wave test tone oscillator somewhere between 40 and 60Hz, and plug it into a mixer channel module containing the noise gate.
  2. Patch the kick drum into the gate’s key input and set the threshold relatively high, so that the kick exceeds the threshold for only a very short amount of time.
  3. Set the noise gate decay for the desired amount of oscillator decay. Hopefully your gate decay can go up to about 2 seconds, but even 1 second can do the job.

Now whenever the kick drum hits, it opens up the gate for a fraction of a second and lets through the sine wave; the decay time then provides the desired fadeout.

Real Time Manipulation

This real-time performance tip can sound very cool with hip-hop, techno, and other types of music that rely on variations within drum loops. With most loops, the snare and kick will reach the highest levels, with (typically) hi-hat below that and percussion (maracas, shakers, tambourine, etc.) mixed in the background. Tweaking the noise gate threshold in real time causes selected parts of the loop to drop out. For example, with the threshold at minimum, you hear the entire loop. Move the threshold up, and the percussion disappears. Move it up further, and the high-hat drops out. Raise it even higher, and the snare and kick lose their decays and become ultra-percussive.

To read the full detailed article with sound samples see: Noise Gates

 

February 4, 2011

Dynamics Processing Meets Rock Guitar: How to Compress a Guitar or Bass

Dynamics processing with studio-oriented processors? Been there, done that. But have you re-visited it lately in a guitar context? Dynamics control for vocals or program material is very different compared to guitar. Much of this is because there are many ways to use dynamics processing for guitar (or bass). So, let’s take a look at the different ways to apply dynamics, with examples of suggested settings.

For an introduction to compression, check out the article “Compressors Demystified.” If you’re already up to speed, let’s give a few basics on how to set up studio processors with guitar (however, note that these same basic techniques work with plug-in software compressors as well as hardware).

The Interface Space

“Stomp box” dynamics processors, while designed specifically for guitar, are more limited than rack-mount studio hardware – but the latter have issue levels with guitar. Interfacing involves one of four approaches:

Use the instrument input. If the processor has an “instrument” input, you’re golden. Plug the guitar directly into the processor, then run it into the mixer, amp modeler, guitar amp (assuming you can adjust the output level to avoid total overload), or whatever. Look for an instrument input impedance above 100kilohms, and preferably above 220kilohms, to avoid dulling high frequencies and reducing level. But too high an impedance (in the 5-10Megohm range) reaches a point of diminishing returns, because now the input may be too sensitive and prone to noise pickup. A 1Megohm impedance is a good compromise setting.

Use a preamp or suitable direct box. Adding a preamp or direct box (assuming it has an appropriately high input impedance) before the processor will preserve the guitar signal’s fidelity and allow for best level matching. If you’re driving a guitar amp, you may be able to use the dynamics processor’s output control to add some extra overdrive, but don’t go overboard (or do, if you like really nasty sounds!).

Insert into your guitar amp’s effects loop. If you want to record with your guitar amp but are using a line-level processor, patch it into the guitar amp’s effects loop. The loop should be able to provide line levels for the send (goes into the processor’s input) and return (comes from the processor’s output).

If you’re using a hardware mixer, insert the dynamics processor into your mixer’s channel inserts. This will also match levels properly, although you’ll still have to figure out how to interface the guitar with the mixer. The choices are the same as above: If the mixer has an instrument input, great. If not, use a preamp, direct box, etc. between the guitar and mixer.

Now let’s take a closer look how to really do it…

Double Your Pleasure

Patching two compressors in series, with both set for small amounts of compression, can give a significant amount of compression but sound less obvious than using a single compressor to give the same amount of compression. The first stage essentially “pre-conditions” the signal so that the second compressor doesn’t have to work so hard.

 

If you have a stereo compressor that can be set to dual mono operation, you can patch the two individual compression channels in series. With plug-ins, you can just insert two in series in a track. The drawback is that unlike standard compression, where you have to adjust only one set of controls, an ˆ la carte approach requires adjusting both sets of compressor controls. While this might seem like a disadvantage, most of the time you’ll set them to similar settings anyway.

Window Shopping

To get an idea of what’s out there in compressor-land, visit a few retailers and manufacturers and you’ll see the choices are huge, ranging from under a hundred dollars to thousands (and thousands!) of dollars. But realistically, for the type of application we’re describing here, you don’t need anything too fancy – it’s not like you’re using the compressor to re-master vintage recordings for audiophile releases. Besides, these days technology is at a level where even fairly inexpensive devices can deliver excellent results.

 

In any event, all the above tips are just guidelines. Experiment with your dynamics processor, and you may find yet another way to exploit these perhaps unglamorous, but extremely useful, devices.

To read the full detailed article see:  How to Compress a Guitar or Bass

July 14, 2009

Trinnov – Optimizer ST

Trinnov Audio presents their new stereo loudspeaker processor, the Optimizer ST which is designed to “take the room out of the acoustic equation and improve the accuracy and consistency of your monitoring system”.

To see more exclusive video demos visit Audiofanzine Videos.

June 15, 2009

EQ and Compression Techniques for Vocals and Acoustic Guitar

As an engineer/producer, one of my biggest early challenges was getting my mixes to sound as polished and balanced as the mixes of songs on my favorite albums. Living in Nashville, I knew the problem wasn’t the players (some of whom had even played on those same favorite albums). I also knew that I was happy enough with the sounds I was recording because when I’d solo a particular track, I liked what I heard. The problem, in a nutshell, was getting all the parts of my mix to fit nicely together. What I’ve learned over time and will describe below are a few simple compression and EQ techniques for vocals and the acoustic guitar in your mixes. These techniques, when used properly, will go a long way towards allowing the vocals and acoustic guitars in your mixes to effectively share the sonic space.

Compression

When I first started reading about compressors I was hopelessly lost. The terminology was technical in an almost mean-spirited way and I couldn’t make heads or tails of what was being written. To keep things simple, I think of compression as a way of evening out the loud and soft parts of any vocal or instrument so that its behavior is a bit more predictable. In other words, compression brings up the really soft spots and tames the really loud spots so that you’re not constantly reaching for the volume fader on your mixing board (or virtual mixing board on your DAW). In its simplest form, a compressor, whether a hardware unit or a plug-in, will squeeze the audio so that its highs and lows are less pronounced. This allows you to do things like bring down the volume level of the compressed track without fear that its softer parts will get lost, or bring up the volume level without fear that the loud parts will jump out. It might help to think of all compression settings (attack, release, ratio and threshold) as ways to squeeze your audio more or less aggressively. Not enough compression will leave tracks that jump out of a mix at inappropriate times or get lost in the sound of the other instruments; however, too much compression can make a track sound lifeless or uninspired. My rule of thumb is to be less aggressive compressing audio on the way into your DAW (because you’re stuck with whatever you do) and more aggressive with my plug-in compression (because you can always dial it back).

EQ

While a wonderful (and essential) tool, EQ is also quite possibly the quickest way to royally mess up the sound of a mix. Overuse of EQ ranks second only to overuse of reverb as the hallmark of an inexperienced mix engineer. EQ should be used to subtly (or not so subtly) color the sound of the particular track you’re working on so that it relates well to and leaves space for the other tracks in a mix. My experience has been that it’s what you pull out and not what you put in that makes EQ work best. For example, even when you’re looking for a boost in the high frequencies of a track, it’s often more effective to pull a few dB from a lower frequency which, in turn, brightens the sound.

Conclusion
Compression and EQ are two very powerful weapons in your mix arsenal, but as with anything, overuse will do more harm than good. I think back to the words of an engineer whose work I really respect who liked to say “I’ll compress until it sucks and then back it off from there.” In other words, knowing when to say “when” is an equally useful skill. A final thought…as far as signal path is concerned, I tend to place compression after EQ because EQ effectively raises or lowers the volume of the track and I’ve found I get a more effective response from the compressor if I hit it with the EQed audio. I would highly recommend using the above EQ and compression settings not as an ironclad rule but rather as a jumping off point. Every mix is different and your ears will tell you what’s working and what isn’t.

To read the full detailed article see: EQ & Compression Techniques

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